Abstract: An apparatus for encoding an audio signal to obtain an encoded audio signal to be used by a decoder having a high frequency reconstruction module for performing a high frequency reconstruction for a frequency range above a crossover frequency includes, a core encoder for encoding a lower frequency band of the audio signal up to the crossover frequency, the crossover frequency being variable, and the core encoder being operable on a block-wise frame by frame basis, and a crossover frequency control module for estimating, dependent on a measure of the degree of difficulty for encoding the audio signal by the core encoder and/or a boarder between a tonal and a noise-like frequency range of the audio signal, the crossover frequency to be selected by the core encoder for a frame of a series of subsequent frames, so that the crossover frequency is variable adaptively over time for the series of subsequent frames.

Abstract: The invention concerns a method and apparatus for performing packet loss or Frame Erasure Concealment (FEC) for a speech coder that does not have a built-in or standard FEC process. A receiver with a decoder receives encoded frames of compressed speech information transmitted from an encoder. A lost frame detector at the receiver determines if an encoded frame has been lost or corrupted in transmission, or erased. If the encoded frame is not erased, the encoded frame is decoded by a decoder and a temporary memory is updated with the decoder's output. A predetermined delay period is applied and the audio frame is then output. If the lost frame detector determines that the encoded frame is erased, a FEC module applies a frame concealment process to the signal. The FEC processing produces natural sounding synthetic speech for the erased frames.

Abstract: A speech coding system is provided with time-domain noise attenuation. The speech coding system has an encoder operatively connected to a decoder via a communication medium. A preprocessor processes a digitized speech signal from an analog-to-digital converter. Speech coding systems are used to encode and decode a bitstream. Gains from the speech coding are adjusted by a gain factor Gf that provides time-domain background noise attenuation.

Abstract: In a method for determining speech quality using an objective measure, in order to enhance prediction reliability of the evaluated quality parameters, distortions of the mean spectral envelope are extensively corrected with a weighting function WT(f) before comparing spectral properties. Additionally, the fixed band limits for integration of spectral power density are suppressed and other band limits are searched for instead in a predetermined optimization area in which the resulting spectral intensity representations of the voice signal to be evaluated and the reference voice signal have maximum similarity. The solutions described can supplement known methods and can be incorporated into their structures.

Abstract: A speech processing system is provided which is operable to receive sets of signal values representative of a speech signal generated by a speech source. The system is operable to determine a measure of the quality of the speech signal by performing a statistical analysis of the received sets of signal values. The system stores data defining a predetermined function derived from a signal model which models the speech source and which defines a probability density function which gives, for a given set of model parameters, the probability that the signal model has those model parameters given that the signal model is assumed to have generated the received set of signal values. The system applies a current set of received signal values to the stored probability density function and then draws samples from it using a Gibbs sampler.

Abstract: An audio test system for analyzing and quantifying audio data losses during network-based telephony sessions between communication devices such as telephony-enabled computers and Internet telephones. A transmit device converts an input audio signal to a stream of data packets and communicates the data stream over a network to a receive device. The receive device converts the data stream to an output audio signal. An audio analyzer is coupled to the transmit device and the receive device to monitor and capture the input audio signal and the output audio signal. The audio analyzer determines transmission qualities for the session, such as data loss and latency, by generating and comparing envelope waveforms of the input audio signal and the output audio signal.

Abstract: A method and apparatus for speaker recognition is provided that matches the noise in training data to noise in testing data using spectral addition. Under spectral addition, the mean and variance for a plurality of frequency components are adjusted in the training data and the test data so that each mean and variance is matched in a resulting matched training signal and matched test signal. The adjustments made to the training data and test data add to the mean and variance of the training data and test data instead of subtracting from the mean and variance.

Abstract: The invention concerns a method and apparatus for performing packet loss or Frame Erasure Concealment (FEC) for a speech coder that does not have a built-in or standard FEC process. A receiver with a decoder receives encoded frames of compressed speech information transmitted from an encoder. A lost frame detector at the receiver determines if an encoded frame has been lost or corrupted in transmission, or erased. If the encoded frame is not erased, the encoded frame is decoded by a decoder and a temporary memory is updated with the decoder's output. A predetermined delay period is applied and the audio frame is then output. If the lost frame detector determines that the encoded frame is erased, a FEC module applies a frame concealment process to the signal. The FEC processing produces natural sounding synthetic speech for the erased frames.

Abstract: The invention concerns a method and apparatus for performing packet loss or Frame Erasure Concealment (FEC) for a speech coder that does not have a built-in or standard FEC process. A receiver with a decoder receives encoded frames of compressed speech information transmitted from an encoder. A lost frame detector at the receiver determines if an encoded frame has been lost or corrupted in transmission, or erased. If the encoded frame is not erased, the encoded frame is decoded by a decoder and a temporary memory is updated with the decoder's output. A predetermined delay period is applied and the audio frame is then output. If the lost frame detector determines that the encoded frame is erased, a FEC module applies a frame concealment process to the signal. The FEC processing produces natural sounding synthetic speech for the erased frames.

Abstract: The invention concerns a method and apparatus for performing packet loss or Frame Erasure Concealment (FEC) for a speech coder that does not have a built-in or standard FEC process. A receiver with a decoder receives encoded frames of compressed speech information transmitted from an encoder. A lost frame detector at the receiver determines if an encoded frame has been lost or corrupted in transmission, or erased. If the encoded frame is not erased, the encoded frame is decoded by a decoder and a temporary memory is updated with the decoder's output. A predetermined delay period is applied and the audio frame is then output. If the lost frame detector determines that the encoded frame is erased, a FEC module applies a frame concealment process to the signal. The FEC processing produces natural sounding synthetic speech for the erased frames.

Abstract: The invention provides a Hidden Markov Model (132) based automated speech recognition system (100) that dynamically adapts to changing background noise by detecting long pauses in speech, and for each pause processing background noise during the pause to extract a feature vector that characterizes the background noise, identifying a Gaussian mixture component of noise states that most closely matches the extracted feature vector, and updating the mean of the identified Gaussian mixture component so that it more closely matches the extracted feature vector, and consequently more closely matches the current noise environment. Alternatively, the process is also applied to refine the Gaussian mixtures associated with other emitting states of the Hidden Markov Model.

Abstract: A method and apparatus estimate additive noise in a noisy signal using an iterative technique within a recursive framework. In particular, the noisy signal is divided into frames and the noise in each frame is determined based on the noise in another frame and the noise determined in a previous iteration for the current frame. In one particular embodiment, the noise found in a previous iteration for a frame is used to define an expansion point for a Taylor series approximation that is used to estimate the noise in the current frame.

Abstract: A suppression system of background noise of speech signals uses an adaptive filter of long-time and short-time statistical characteristics of the speech signals. Since the statistical characteristics of the speech signals vary with time, the associated coefficents of the filter also have to be adjusted according to the varitation of the speech signals to eliminate the unnecessary background noise. High frequency attenuation of the speech signals is compensated for by passing the signal through a high frequency booster to elevate the degree of brightness of the speech signals and to improve their quality.

Abstract: A scalable audio codec processes, quantizes and encodes audio signals into an embedded audio bitstream of bit-planes each having a data unit. The data unit has a beginning refinement bits partition, a second significance bits partition, a third sign boundary mark bits partition, and a fourth sign bits partition. The second and fourth partitions form a boundary for the third partition. The quantizing uses a variable length coding algorithm. The third partition is an invalid codeword for a predetermined encoding method being used to encode. The codec uses a decoder to decode the embedded audio bitstream of bit-planes using Reversible exponential Golomb (Exp-Golomb) codes in a Reversible Variable Length Code (RVLC) algorithm to produce quantized data of weighted subbands. An inverse quantizer dequantizes the quantized data into audio signals.

Abstract: A method, apparatus and program product facilitates the sharing of memory resources between exclusive audio post-processes. A program identifies post-processing applications that execute at different instances and assigns to them a common memory block. An audio packet arrives at a digital signal processor (DSP). The DSP associates a frame of the packet with a post-process. The DSP buffers the frame to a memory block that corresponds to the post-process. Upon releasing the buffered frame, the DSP prepares the memory block for use with a second post-process and frame.

Abstract: In a digital channel of a digital wireless communication system including at least one mobile station, at least one base transceiver station in communication with the mobile station, a transcoder configured to provide a signal conversion between vocoder frames and pulse code modulation, and a mobile switching center for interconnecting the digital wireless communication system to a public switched telephone network, a method and apparatus for determining a fault in the digital channel is disclosed. The method includes generating a first set of vocoder input parameters from a speech input signal, and generating a second set of vocoder input parameters from an output signal substantially equivalent to the speech input signal as it is received at a mobile station via the digital channel. The method further includes calculating a metric based on the first and the second set of vocoder input parameters, and subsequently determining a fault in the digital channel using the metric.

Abstract: A method and system for calibration of a data acquisition path is achieved by applying a voice utterance to a first high quality microphone and reference path and to a test acquisition path including a test microphone such as a lower quality one used in a car. The calibration device includes detecting the power density of the reference signal YR through the reference path and detecting the power density of the signal YN through the acquisition path. A processor processes these signals to provide an output signal representing a noise estimate and channel estimate. The processing uses equation derived by modeling convolutive and additive noise as polynomials with different orders and estimating model parameters using maximum likelihood criterion and simultaneously solving linear equations for the different orders.

Abstract: Pursuant to one aspect of the invention, a prefilter module that incorporates an inverse filter is used in conjunction with an encoder. The inverse filter has an inverse frequency response of a frequency response of a filter that simulates speech having transmission path characteristics, such as telephone-channel bandwidth speech, and/or noisy speech. The inverse filter is used to compensate transmission path characteristics of an input signal. The inverse filter can be designed using several methods, such as, for example, an autoregressive model or a moving average model. Pursuant to a second aspect of the invention, a parameter preprocessor is used in conjunction with a decoder. The parameter preprocessor performs pitch rectification through use of a medium and linear filter, and updates spectral amplitudes and voicing parameter depending on the pitch rectification.

Abstract: A method, apparatus and system that receives speech commands at a remote control device, digitizes those speech commands, and transmits the digitized speech commands to an electronic device, such as a digital home communication terminal (DCHT). The electronic device interprets the speech commands to allow the remote control operator to control the electronic device.

Abstract: A method of concealing bit errors in a signal is provided. The method comprises encoding a signal parameter according to a set of constraints placed on a signal parameter quantizer. The encoded signal parameter is decoded and compared against the set of constraints. Finally, the method includes declaring the decoded signal parameter invalid when the set of constraints is violated. Training binned ranges of gain values provide a threshold for selecting data segments to examine for violation of constraints on gain differences. Further, an additional method comprises training a threshold function T(qlg(m?1), ?qlg(m?1) used in a codec bit error detecting technique. The threshold function is based upon a first training file having N signal segments. The method includes encoding the first training file and determining gain values qlg(m) of each of the N signal segments within the encoded first training file. The gain values form a range and the range is divided into bins.

Abstract: At an audio source, pause information is added to audio data, the combination of which is subsequently packetized. The resulting packets are transmitted to an audio destination via a network in which different packets may be subjected to varying levels of delay. At the audio destination, the pause information may be used to insert pauses at appropriate times to accommodate the occurrence of delays in packet delivery. In one embodiment, pauses are inserted based on a hierarchy of pause types. During pauses, audio filler information may be injected. In this manner, the effects of variable network delays upon reconstructed audio may be mitigated.

Abstract: An application server generates and maintains a server-side data record, also referred to as a “brownie”, that includes application state information and user attribute information for multiple users within a single session controlled by a web-based browser. The brownie includes a session identifier that uniquely identifies the session, and a subsession identifier that uniquely identifies each corresponding user of the application session. As each new user is added to the session, for example by initiating a call to the new user, the application server stores the subsession identifier and corresponding application state information for the new user in the same brownie. In response to receiving a second web page request from the browser that includes the session identifier, the application server initiates a new web application instance, and recovers the brownie from the memory based on the session identifier included in the second page request.

Abstract: An input signal enters a noise suppression system in a time domain and is converted to a frequency domain. The noise suppression system then estimates a signal to noise ratio of the frequency domain signal. Next, a signal gain is calculated based on the estimated signal to noise ratio and a voicing parameter. The voicing parameter may be determined based on the frequency domain signal or may be determined based on a signal ahead of the frequency domain signal with respect to time. In that event, the voicing parameter is fed back to the noise suppression system, for example, by a speech coder, to calculate the signal gain. After calculating the gain, the noise suppression system modifies the signal using the calculated gain to enhance the signal quality. The modified signal may further be converted from the frequency domain back to the time domain for speech coding.

Abstract: A background sound sending side multiplexes and sends, in a multiplexer, uttered encoded speech data generated in a speech sending section and encoded background sound data outputted from a background sound storing section. Simultaneously, a background sound reproducing section, reproduces encoded background sound data and reproduced background sound signal is superposed on received speech in a receiving section and outputted from a receiver. A background sound receiving side demultiplexes, in a demultiplexer, received multiplexed data into received encoded speech data and encoded background sound data which are decoded in the receiving section and the background sound reproducing section respectively, and in the receiving section, a sound in which received speech and background sound are superposed is outputted from a receiver.

Abstract: A decoding processing portion 11 of a speech decoder 10 is provided with an emphasis processing portion 15 for performing an emphasis process on signals to be processed (excited signals) SPC generated from coded speech signals BS. A counter portion 17 counts the number of times code errors occurred in successive frames of the coded speech signal BS, and outputs the successive frame error number. When the successive frame error number outputted form the counter portion 17 is less than or equal to a preset reference successive frame error number, a first switch SW1 and second switch SW2 are set to an emphasis processing portion 15 side. Accordingly, the signals to be processed SPC generated from various parameters included in the coded speech signals are supplied through the switch SW1 to the emphasis processing portion 15 of the decoding processing portion 11 to perform an emphasis process.

Abstract: A system and method of masking an objectionable artifact in a mobile telephone. The process starts by referring to the design characteristics of the mobile phone to determine the expected level and source of objectionable artifacts that will be apparent during operation of the mobile phone. The noise necessary to mask the objectionable artifact is then calculated. The masking noise signal is then created within the mobile telephone and superimposed over the objectionable artifact. The masking noise signal is typically created in a digital signal processor (DSP) resident within the mobile telephone using a pseudo-noise generator. The masking noise signal can be stored as a look-up table in the digital signal processor (DSP) of the mobile telephone. To further enhance the effectiveness of the masking process, the masking noise signal is filtered to best match the objectionable artifact. The masking noise signal can be low pass filtered for objectionable artifacts that are lower in frequency.

Abstract: A handheld audio spectrum analyzer includes a stored STI measurement algorithm and a selector for selecting the stored STI measurement algorithm to process a transduced sound signal received by a microphone associated with the handheld audio spectrum analyzer to provide the STI between the microphone and the source of the sound signal that transduces an audio test signal related to the STI measurement algorithm stored in the handheld audio spectrum analyzer.

Abstract: The data stream between the transcoders (TCE1, TCE2) of a mobile wireless system is subdivided into a first data stream with samples for transmission and a second data stream with signal parameters for reconstruction of user data and/or for signaling. Both data streams are transmitted at the same time in particular in a handshake phase. The invention permits an improvement in the quality of transmitted data, e.g., speech data in a GSM network in tandem operation between mobile subscribers, in particular during a handshake phase.

Abstract: A digital audio reproducing apparatus including a receiver receiving modulated data, a demodulator demodulating the modulated data received by the receiver, an audio decoder decoding, in a unit of a frame, digital audio information contained in the modulated data demodulated by the demodulator, and an audibility corrector for effecting audibility correction on failing digital audio information contained in a frame that failed to be decoded, when the audio decoder fails to decode the digital audio information.

Abstract: An automated speech recognition filter is disclosed. The automated speech recognition filter device provides a speech signal to an automated speech platform that approximates an original speech signal as spoken into a transceiver by a user. In providing the speech signal, the automated speech recognition filter determines various models representative of a cumulative signal degradation of the original speech signal from various devices along a transmission signal path and a reception signal path between the transceiver and a device housing the filter. The automated speech platform can thereby provide an audio signal corresponding to a context of the original speech signal.

Abstract: In a Noise Feedback Coding (NFC) system having a corresponding ZERO-STATE filter structure, the first ZERO-STATE filter structure including multiple filters, a method of producing a ZERO-STATE response error vector. The method includes: (a) transforming the first ZERO-STATE filter structure to a second ZERO-STATE filter structure including only an all-zero filter, the all-zero filter having a filter response substantially equivalent to a filter response of the ZERO-STATE filter structure including multiple filters; and (b) filtering a VQ codevector with the all-zero filter to produce the ZERO-STATE response error vector corresponding to the VQ codevector.

Abstract: A audio decoding device includes an error correction processing means 13 for generating interpolation process information, a soft decision information generating means 12a for generating a soft decision information indicating a current situation of the transmission line based on coded data, a audio decoding processing means 14 for applying an interpolation process to the audio code in unit of bit based on interpolation information and soft decision information and then decoding the audio code which is subjected to the interpolation process to generate audio data, and a audio output processing means 15 for outputting the audio data.

Abstract: A speech decoder 10 comprises a decoding processing portion 11 and an amplification process control portion 12. Here, the decoding processing potion 11 is a device for decoding a received coded speech signal (bitstream) BS and outputting a decoded speech signal SP. Additionally, the amplification process control portion 12 monitors the state of occurrence of frame errors in the coded speech signal BS, and when the number of successive frame errors exceeds a predetermined reference frame error number, outputs amplification instructions for a predetermined number of frames after the successive frame errors disappear. As a result, instead of codebook data DCB obtained by a decoding process of the decoding processing portion 11, amplified codebook data DACB are supplied to a synthesis filter portion 17, and is written into the codebook decoder 18 of the decoding processing portion 11 as new original codebook data DCBO.

Abstract: In a voice communication system having a transmitter and receiver on opposing sides of an interface, the transmitter is switched on to transmit speech components and is switched off during speech pauses. To provide comfort noise at the receiver, and thus avoid annoying effects caused by continual switching of the transmitter, a comfort noise generator disposed to produce comfort noise of an adjustable amplitude is located on the receiver side of the interface. A first subsystem responsive to operation of the transmitter provides a flag to the receiver to commence operation of the comfort noise generator, when the transmitter discontinues transmission in response to a speech pause. A second subsystem transmits a succession of amplitude parameters through the interface to selectively adjust the amplitude of the generator in corresponding relationship with a noise level at the transmitter.

Abstract: A method and apparatus are provided for generating comfort noise in a communication device. The method includes receiving a signal, scaling the signal to a preselected value, indicating whether an error occurred during transmission of the signal, and providing the scaled signal as an output signal in response to receiving the indication that the error occurred during transmission. The apparatus includes a scaler for receiving a signal and being capable of scaling the signal to a preselected value. The apparatus includes an indicator capable of indicating that an error occurred during transmission of the signal, wherein the scaled signal is provided as an output signal in response to an indication that the error occurred during transmission.

Abstract: A method and system for encoding and decoding an input signal, wherein the input signal is divided into a higher frequency band and a lower frequency band in the encoding and decoding processes, and wherein the decoding of the higher frequency band is carried out by using an artificial signal along with speech related parameters obtained from the lower frequency band. In particular, the artificial signal is scaled before it is transformed into an artificial wideband signal containing colored noise in both the lower and the higher frequency band. Additionally, voice activity information is used to define speech periods and non-speech periods of the input signal. Based on the voice activity information, different weighting factors are used to scale the artificial signal in speech periods and non-speech periods.

Abstract: A method of creating a compressed audio output signal from a series of input audio signals is disclosed and claimed. In one embodiment, the method may include, for each of the input audio signals a) precomputing a transform corresponding to the desired compression format of the output audio signal. This may be followed by b) precomputing ancillary information relating to the compression of the transformed input audio. Next, the method may include c) mixing together the transformed input signals in the transform domain to produce an output transform domain signal. The method may then include d) algorithmically combining together the precomputed ancillary information to determine a suitable decompression strategy. Lastly, the method may include e) outputting compressed audio data comprising the output transform domain signal and the combined ancillary information.

Abstract: A digital audio receiver stores received frames temporarily for decoding and error concealment. A reconstructing block (14) in the decoder reads stored frames using a read window (43) wherein the latest received frame (+cnnxt) is undecoded. Decoding is carried out in stages so that the correctness of the current frame (0) is examined and possible errors are concealed using corresponding data of other frames in the window. Detection of errors is based on checksums (19, 26) and allowed values of bit combinations in certain parts of the frame. In addition, the receiver maintains an estimate (60) for the signal's bit error ratio and uses it to control the operation of the error concealment algorithm.

Abstract: The invention describes a system that generates a wide band signal (100-7000 Hz) from a telephony band (or narrow band: 300-3400 Hz) speech signal to obtain an extended band speech signal (100-3400 Hz). This technique is particularly advantageous since it increases signal naturalness and listening comfort with keeping compatibility with all current telephony systems. The described technique is inspired on Linear Predictive speech coders. The speech signal is thus split into a spectral envelope and a short-term residual signal. Both signals are extended separately and recombined to create an extended band signal.

Abstract: A method and system for providing comfort noise in the non-speech periods in speech communication. The comfort noise is generated based on whether the background noise in the speech input is stationary or non-stationary. If the background noise is non-stationary, a random component is inserted in the comfort noise using a dithering process. If the background noise is stationary, the dithering process is not used.

Abstract: A source signal (e.g. a speech sample) is processed or transmitted by a speech coder 1 and converted into a reception signal (coded speech signal). The source and reception signals are separately subjected to preprocessing 2 and psychoacoustic modelling 3. This is followed by a distance calculation 4, which assesses the similarity of the signals. Lastly, an MOS calculation is carried out in order to obtain a result comparable with human evaluation. According to the invention, in order to assess the transmission quality a spectral similarity value is determined which is based on calculation of the covariance of the spectra of the source signal and reception signal and division of the covariance by the standard deviations of the two said spectra. The method makes it possible to obtain an objective assessment (speech quality prediction) while taking the human auditory process into account.

Abstract: An adaptive noise suppression system includes an input A/D converter, an analyzer, a filter, and a output D/A converter. The analyzer includes both feed-forward and feedback signal paths that allow it to compute a filtering coefficient, which is input to the filter. In these paths, feed-forward signal are processed by a signal to noise ratio estimator, a normalized coherence estimator, and a coherence mask. Also, feedback signals are processed by a auditory mask estimator. These two signal paths are coupled together via a noise suppression filter estimator. A method according to the present invention includes active signal processing to preserve speech-like signals and suppress incoherent noise signals. After a signal is processed in the feed-forward and feedback paths, the noise suppression filter estimator then outputs a filtering coefficient signal to the filter for filtering the noise out of the speech and noise digital signal.

Abstract: A speech decoding unit estimates coding parameters of a speech pause by carrying out smoothing algorithm of the coding parameters by using a coding parameter xref constituting far-end talker background noise information extracted by a parameter extracting circuit 12, and a coding parameter xn used for synthesizing the previous background noise.

Abstract: A method is provided for generating comfort noise in a packetized voice communication system having a transmitter and a receiver. The receiver is provided with a buffer for storing voice packets. The buffer is chosen to be of a predetermine size such that, upon halting the transmitter as a result of silence detection, the buffer is filled with actual silence samples from the transmitter. A comparator compares an output TDM sample pointer with a start of silence pointer of the buffer. In the event that the pointers are the same, silence is flagged and a random number generator loads numbers into the TDM sample pointer for outputting a random sequence of the silence packets to the telephone receiver.

Abstract: Reducing mismatch between HMMs trained with clean speech and speech signals recorded under background noise can be approached by distribution adaptation using parallel model combination (PMC). Accurate PMC has no closed-form expression, therefore simplification assumptions must be made in implementation. Under a new log-max assumption, adaptation formula for log-spectral parameters are presented, both for static and dynamic parameters.

Abstract: A method and apparatus are provided for reducing noise in a signal. Under one aspect of the invention, a correction vector is selected based on a noisy feature vector that represents a noisy signal. The selected correction vector incorporates dynamic aspects of pattern signals. The selected correction vector is then added to the noisy feature vector to produce a cleaned feature vector. In other aspects of the invention, a noise value is produced from an estimate of the noise in a noisy signal. The noise value is subtracted from a value representing a portion of the noisy signal to produce a noise-normalized value. The noise-normalized value is used to select a correction value that is added to the noise-normalized value to produce a cleaned noise-normalized value. The noise value is then added to the cleaned noise-normalized value to produce a cleaned value representing a portion of a cleaned signal.

Abstract: A decoding circuit, for receiving a bit stream including an encoded audio signal and header information used for-decoding the encoded audio signal, and decoding the encoded audio signal based on the header information, includes a header analysis section for outputting at least one decoding parameter obtained from the header information and decoding parameter change information indicating whether or not the at least one decoding parameter has been changed; a signal processing section for decoding the encoded audio signal, based on the at least one decoding parameter, into a decoded signal and outputting the decoded signal; an automatic mute processing section for executing automatic mute on the decoded signal after the at least one decoding parameter is changed; and an output section for outputting the decoded signal output from the automatic mute processing section.

Abstract: A method for calculating a postfilter frequency response for filtering digitally processed speech, the method comprising identifying at least one format of a speech spectrum of the digitally processed speech; and normalizing points of the speech spectrum with respect to an identified format.

Abstract: A comfort noise generating apparatus designed to reduce a feeling of fracture in a voice sound and a feeling of unsuitableness at timings before and after a process by an NLP or the like. A comfort noise according to a voice signal after an unnecessary component has been removed is formed and added to this voice signal. The apparatus comprises: a noise generator for generating a noise which is multiplexed to a voice signal after completion of a process by an NLP; a signal level analyzing part for measuring a signal power of a voice signal before the process; a multiplier for varying noise characteristics from the noise generator in accordance with an analysis result of the signal level analyzing part; a signal frequency characteristics analyzing part for selecting the optimum coefficient from a coefficient register; a band pass filter for forming a comfort noise by the selected coefficient and a white noise signal; and an adder for multiplexing the comfort noise to the voice signal.

Abstract: A mapping function is generated between subjective measures of audio signal quality, e.g., mean opinion score (MOS) or degradation MOS (DMOS) measures, and corresponding objective distortion measures, e.g., auditory speech quality measures (ASQMs) or perceptual speech quality measures (PSQMs), for known audio signals. The subjective measures and corresponding objective distortion measures are determined in accordance with modulated noise reference unit (MNRU) conditions or other suitable distortion conditions placed on the source speech, and a regression analysis is applied to the results to generate the mapping function. The mapping function may then be utilized, e.g., to evaluate speech quality of additional source speech from a particular speech coding system. In this case, the objective distortion measure is generated using the additional source speech, and the resulting objective measure is applied as an input to the mapping function to generate an estimate of the value of the subjective measure.